NAME

sound, pcm, snd -- FreeBSD PCM audio device infrastructure

SYNOPSIS

To compile this driver into the kernel, place the following line in your
kernel configuration file:
devicesound
Non-PnP sound cards require the following lines in device.hints(5):
hint.pcm.0.at="isa"
hint.pcm.0.irq="5"
hint.pcm.0.drq="1"
hint.pcm.0.flags="0x0"

DESCRIPTION

The sound driver provides support for PCM audio play and capture. This
driver also supports various PCI, ISA, WSS/MSS compatible sound cards,
AC97 mixer and High Definition Audio. Once the sound driver attaches,
supported devices provide audio record and playback channels. The
FreeBSD sound system provides dynamic mixing ``VCHAN'' and rate
conversion ``soft formats''. True full duplex operation is available on
most sound cards.
If the sound card is supported by a bridge driver, the sound driver works
in conjunction with the bridge driver.
Apart from the usual parameters, the flags field is used to specify the
secondary DMA channel (generally used for capture in full duplex cards).
Flags are set to 0 for cards not using a secondary DMA channel, or to
0x10 + C to specify channel C.
The driver does its best to recognize the installed hardware and drive it
correctly so the user is not required to add several lines in
/boot/device.hints. For PCI and ISA PnP cards this is actually easy
since they identify themselves. For legacy ISA cards, the driver looks
for MSS cards at addresses 0x530 and 0x604 (unless overridden in
/boot/device.hints).
BootVariables
In general, the module snd_foo corresponds to devicesnd_foo and can be
loaded by the boot loader(8) via loader.conf(5) or from the command line
using the kldload(8) utility. Options which can be specified in
/boot/loader.conf include:
snd_driver_load (``NO'') If set to ``YES'', this option loads all
available drivers.
snd_emu10k1_load (``NO'') If set to ``YES'', only the SoundBlaster
5.1 driver and dependent modules will be loaded.
snd_foo_load (``NO'') If set to ``YES'', load driver for
card/chipset foo.
To define default values for the different mixer channels, set the
channel to the preferred value using hints, e.g.: hint.pcm.0.line="0".
This will mute the input channel per default.
MultichannelAudio
Multichannel audio, popularly referred to as ``surround sound'' is
supported and enabled by default. The FreeBSD multichannel matrix
processor supports up to 18 interleaved channels, but the limit is
currently set to 8 channels (as commonly used for 7.1 surround sound).
The internal matrix mapping can handle reduction, expansion or re-routing
of channels. This provides a base interface for related multichannel
ioctl() support. Multichannel audio works both with and without VCHANs.
Most bridge device drivers are still missing multichannel matrixing
support , but in most cases this should be trivial to implement. Use the
dev.pcm.%d.[play|rec].vchanformatsysctl(8) to adjust the number of
channels used. The current multichannel interleaved structure and
arrangement was implemented by inspecting various popular UNIX
applications. There were no single standard, so much care has been taken
to try to satisfy each possible scenario, despite the fact that each
application has its own conflicting standard.
EQ
The Parametric Software Equlizer (EQ) enables the use of ``tone''
controls (bass and treble). Commonly used for ear-candy or frequency
compensation due to the vast difference in hardware quality. EQ is
disabled by default, but can be enabled with the hint.pcm.<X>.eq tunable.
VCHANs
Each device can optionally support more playback and recording channels
than physical hardware provides by using ``virtual channels'' or VCHANs.
VCHAN options can be configured via the sysctl(8) interface but can only
be manipulated while the device is inactive.
VPC
FreeBSD supports independent and individual volume controls for each
active application, without touching the master sound volume. This is
sometimes referred to as Volume Per Channel (VPC). The VPC feature is
enabled by default.
LoaderTunables
The following loader tunables are used to set driver configuration at the
loader(8) prompt before booting the kernel, or they can be stored in
/boot/loader.conf in order to automatically set them before booting the
kernel. It is also possible to use kenv(1) to change these tunables
before loading the sound driver. The following tunables can not be
changed during runtime using sysctl(8).
hint.pcm.<X>.eq
Set to 1 or 0 to explicitly enable (1) or disable (0) the
equalizer. Requires a driver reload if changed. Enabling this
will make bass and treble controls appear in mixer applications.
This tunable is undefined by default. Equalizing is disabled by
default.
hint.pcm.<X>.vpc
Set to 1 or 0 to explicitly enable (1) or disable (0) the VPC
feature. This tunable is undefined by default. VPC is however
enabled by default.
RuntimeConfiguration
There are a number of sysctl(8) variables available which can be modified
during runtime. These values can also be stored in /etc/sysctl.conf in
order to automatically set them during the boot process. hw.snd.* are
global settings and dev.pcm.* are device specific.
hw.snd.compat_linux_mmap
Linux mmap(2) compability. The following values are supported
(default is 0):
-1 Force disabling/denying PROT_EXEC mmap(2) requests.
0 Auto detect proc/ABI type, allow mmap(2) for Linux
applications, and deny for everything else.
1 Always allow PROT_EXEC page mappings.
hw.snd.default_auto
Enable to automatically assign default sound unit to the most
recent attached device.
hw.snd.default_unit
Default sound card for systems with multiple sound cards. When
using devfs(5), the default device for /dev/dsp. Equivalent to a
symlink from /dev/dsp to /dev/dsp${hw.snd.default_unit}.
hw.snd.feeder_eq_exact_rate
Only certain rates are allowed for precise processing. The
default behavior is however to allow sloppy processing for all
rates, even the unsupported ones. Enable to toggle this
requirement and only allow processing for supported rates.
hw.snd.feeder_rate_max
Maximum allowable sample rate.
hw.snd.feeder_rate_min
Minimum allowable sample rate.
hw.snd.feeder_rate_polyphase_max
Adjust to set the maximum number of allowed polyphase entries
during the process of building resampling filters. Disabling
polyphase resampling has the benefit of reducing memory usage, at
the expense of slower and lower quality conversion. Only
applicable when the SINC interpolator is used. Default value is
183040. Set to 0 to disable polyphase resampling.
hw.snd.feeder_rate_quality
Sample rate converter quality. Default value is 1, linear
interpolation. Available options include:
0 Zero Order Hold, ZOH. Very fast, but with poor quality.
1 Linear interpolation. Fast, quality is subject to personal
preference. Technically the quality is poor however, due to
the lack of anti-aliasing filtering.
2 Bandlimited SINC interpolator. Implements polyphase banking
to boost the conversion speed, at the cost of memory usage,
with multiple high quality polynomial interpolators to
improve the conversion accuracy. 100% fixed point, 64bit
accumulator with 32bit coefficients and high precision sample
buffering. Quality values are 100dB stopband, 8 taps and 85%
bandwidth.
3 Continuation of the bandlimited SINC interpolator, with 100dB
stopband, 36 taps and 90% bandwidth as quality values.
4 Continuation of the bandlimited SINC inteprolator, with 100dB
stopband, 164 taps and 97% bandwidth as quality values.
hw.snd.feeder_rate_round
Sample rate rounding threshold, to avoid large prime division at
the cost of accuracy. All requested sample rates will be rounded
to the nearest threshold value. Possible values range between 0
(disabled) and 500. Default is 25.
hw.snd.latency
Configure the buffering latency. Only affects applications that
do not explicitly request blocksize / fragments. This tunable
provides finer granularity than the hw.snd.latency_profile
tunable. Possible values range between 0 (lowest latency) and 10
(highest latency).
hw.snd.latency_profile
Define sets of buffering latency conversion tables for the
hw.snd.latency tunable. A value of 0 will use a low and
aggressive latency profile which can result in possible underruns
if the application cannot keep up with a rapid irq rate,
especially during high workload. The default value is 1, which
is considered a moderate/safe latency profile.
hw.snd.maxautovchans
Global VCHAN setting that only affects devices with at least one
playback or recording channel available. The sound system will
dynamically create up to this many VCHANs. Set to ``0'' if no
VCHANS are desired. Maximum value is 256.
hw.snd.report_soft_formats
Controls the internal format conversion if it is available
transparently to the application software. When disabled or not
available, the application will only be able to select formats
the device natively supports.
hw.snd.report_soft_matrix
Enable seamless channel matrixing even if the hardware does not
support it. Makes it possible to play multichannel streams even
with a simple stereo sound card.
hw.snd.verbose
Level of verbosity for the /dev/sndstat device. Higher values
include more output and the highest level, four, should be used
when reporting problems. Other options include:
0 Installed devices and their allocated bus resources.
1 The number of playback, record, virtual channels, and flags
per device.
2 Channel information per device including the channel's
current format, speed, and pseudo device statistics such as
buffer overruns and buffer underruns.
3 File names and versions of the currently loaded sound
modules.
4 Various messages intended for debugging.
hw.snd.vpc_0db
Default value for sound volume. Increase to give more room for
attenuation control. Decrease for more amplification, with the
possible cost of sound clipping.
hw.snd.vpc_autoreset
When a channel is closed the channel volume will be reset to 0db.
This means that any changes to the volume will be lost. Enabling
this will preserve the volume, at the cost of possible confusion
when applications tries to re-open the same device.
hw.snd.vpc_mixer_bypass
The recommended way to use the VPC feature is to teach
applications to use the correct ioctl(): SNDCTL_DSP_GETPLAYVOL,
SNDCTL_DSP_SETPLAYVOL, SNDCTL_DSP_SETRECVOL,
SNDCTL_DSP_SETRECVOL. This is however not always possible.
Enable this to allow applications to use their own existing mixer
logic to control their own channel volume.
hw.snd.vpc_reset
Enable to restore all channel volumes back to the default value
of 0db.
dev.pcm.%d.bitperfect
Enable or disable bitperfect mode. When enabled, channels will
skip all dsp processing, such as channel matrixing, rate
converting and equalizing. The pure sound stream will be fed
directly to the hardware. If VCHANs are enabled, the bitperfect
mode will use the VCHAN format/rate as the definitive format/rate
target. The recommended way to use bitperfect mode is to disable
VCHANs and enable this sysctl. Default is disabled.
dev.pcm.%d.[play|rec].vchans
The current number of VCHANs allocated per device. This can be
set to preallocate a certain number of VCHANs. Setting this
value to ``0'' will disable VCHANs for this device.
dev.pcm.%d.[play|rec].vchanformat
Format for VCHAN mixing. All playback paths will be converted to
this format before the mixing process begins. By default only 2
channels are enabled. Available options include:
s16le:1.0
Mono
s16le:2.0
Stereo, 2 channels (left, right).
s16le:2.1
3 channels (left, right, LFE).
s16le:3.0
3 channels (left, right, rear center).
s16le:4.0
Quadraphonic, 4 channels (front/rear left and right).
s16le:4.1
5 channels (4.0 + LFE).
s16le:5.0
5 channels (4.0 + center).
s16le:5.1
6 channels (4.0 + center + LFE).
s16le:6.0
6 channels (4.0 + front/rear center).
s16le:6.1
7 channels (6.0 + LFE).
s16le:7.1
8 channels (4.0 + center + LFE + left and right side).
dev.pcm.%d.[play|rec].vchanmode
VCHAN format/rate selection. Available options include:
fixed
Channel mixing is done using fixed format/rate. Advanced
operations such as digital passthrough will not work. Can be
considered as a ``legacy'' mode. This is the default mode
for hardware channels which lack support for digital formats.
passthrough
Channel mixing is done using fixed format/rate, but advanced
operations such as digital passthrough also work. All
channels will produce sound as usual until a digital format
playback is requested. When this happens all other channels
will be muted and the latest incoming digital format will be
allowed to pass through undisturbed. Multiple concurrent
digital streams are supported, but the latest stream will
take precedence and mute all other streams.
adaptive
Works like the ``passthrough'' mode, but is a bit smarter,
especially for multiple sound channels with different
format/rate. When a new channel is about to start, the
entire list of virtual channels will be scanned, and the
channel with the best format/rate (usually the
highest/biggest) will be selected. This ensures that mixing
quality depends on the best channel. The downside is that
the hardware DMA mode needs to be restarted, which may cause
annoying pops or clicks.
dev.pcm.%d.[play|rec].vchanrate
Sample rate speed for VCHAN mixing. All playback paths will be
converted to this sample rate before the mixing process begins.
dev.pcm.%d.polling
Experimental polling mode support where the driver operates by
querying the device state on each tick using a callout(9)
mechanism. Disabled by default and currently only available for
a few device drivers.
RecordingChannels
On devices that have more than one recording source (ie: mic and line),
there is a corresponding /dev/dsp%d.r%d device.
Statistics
Channel statistics are only kept while the device is open. So with
situations involving overruns and underruns, consider the output while
the errant application is open and running.
IOCTLSupport
The driver supports most of the OSS ioctl() functions, and most
applications work unmodified. A few differences exist, while memory
mapped playback is supported natively and in Linux emulation, memory
mapped recording is not due to VM system design. As a consequence, some
applications may need to be recompiled with a slightly modified audio
module. See <sys/soundcard.h> for a complete list of the supported
ioctl() functions.

FILES

The sound drivers may create the following device nodes:
/dev/audio%d.%d Sparc-compatible audio device.
/dev/dsp%d.%d Digitized voice device.
/dev/dspW%d.%d Like /dev/dsp, but 16 bits per sample.
/dev/dsp%d.p%d Playback channel.
/dev/dsp%d.r%d Record channel.
/dev/dsp%d.vp%d Virtual playback channel.
/dev/dsp%d.vr%d Virtual recording channel.
/dev/sndstat Current sound status, including all channels and
drivers.
The first number in the device node represents the unit number of the
sound device. All sound devices are listed in /dev/sndstat. Additional
messages are sometimes recorded when the device is probed and attached,
these messages can be viewed with the dmesg(8) utility.
The above device nodes are only created on demand through the dynamic
devfs(5) clone handler. Users are strongly discouraged to access them
directly. For specific sound card access, please instead use /dev/dsp or
/dev/dsp%d.

DIAGNOSTICS

pcm%d:play:%d:dsp%d.p%d:playinterrupttimeout,channeldead The
hardware does not generate interrupts to serve incoming (play) or
outgoing (record) data.
unsupportedsubdeviceXX A device node is not created properly.

HISTORY

The sound device driver first appeared in FreeBSD 2.2.6 as pcm, written
by Luigi Rizzo. It was later rewritten in FreeBSD 4.0 by Cameron Grant.
The API evolved from the VOXWARE standard which later became OSS
standard.